Remove CodecInst pt.2
The following APIs on AudioCodingModule are deprecated with this CL:
static int NumberOfCodecs();
static int Codec(int, CodecInst*);
static int Codec(const char*, CodecInst*, int, size_t);
static int Codec(const char*, int, size_t);
absl::optional<CodecInst> SendCodec() const;
bool RegisterReceiveCodec(int, const SdpAudioFormat&);
int RegisterExternalReceiveCodec(int, AudioDecoder*, int, int, const std::string&);
int UnregisterReceiveCodec(uint8_t);
int32_t ReceiveCodec(CodecInst*);
absl::optional<SdpAudioFormat> ReceiveFormat();
As well as this method on RtpRtcp module:
int32_t RegisterSendPayload(const CodecInst&);
Bug: webrtc:7626
Change-Id: I1230732136f1fe9048cf74afdeab767ca57ac9ce
Reviewed-on: https://webrtc-review.googlesource.com/c/113816
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26025}diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index c0aab3a..60da679 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -18,7 +18,6 @@
#include "api/array_view.h"
#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
-#include "modules/audio_coding/acm2/rent_a_codec.h"
#include "modules/include/module_common_types.h"
#include "modules/include/module_common_types_public.h"
#include "rtc_base/buffer.h"
@@ -45,9 +44,6 @@
void ModifyEncoder(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)>
modifier) override;
- // Get current send codec.
- absl::optional<CodecInst> SendCodec() const override;
-
// Sets the bitrate to the specified value in bits/sec. In case the codec does
// not support the requested value it will choose an appropriate value
// instead.
@@ -90,19 +86,8 @@
void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
- bool RegisterReceiveCodec(int rtp_payload_type,
- const SdpAudioFormat& audio_format) override;
-
- int RegisterExternalReceiveCodec(int rtp_payload_type,
- AudioDecoder* external_decoder,
- int sample_rate_hz,
- int num_channels,
- const std::string& name) override;
-
// Get current received codec.
- int ReceiveCodec(CodecInst* current_codec) const override;
-
- absl::optional<SdpAudioFormat> ReceiveFormat() const override;
+ absl::optional<std::pair<int, SdpAudioFormat>> ReceiveCodec() const override;
// Incoming packet from network parsed and ready for decode.
int IncomingPacket(const uint8_t* incoming_payload,
@@ -141,8 +126,6 @@
int DisableOpusDtx() override;
- int UnregisterReceiveCodec(uint8_t payload_type) override;
-
int EnableNack(size_t max_nack_list_size) override;
void DisableNack() override;
@@ -317,42 +300,6 @@
}
}
-// Wraps a raw AudioEncoder pointer. The idea is that you can put one of these
-// in a unique_ptr, to protect the contained raw pointer from being deleted
-// when the unique_ptr expires. (This is of course a bad idea in general, but
-// backwards compatibility.)
-class RawAudioEncoderWrapper final : public AudioEncoder {
- public:
- RawAudioEncoderWrapper(AudioEncoder* enc) : enc_(enc) {}
- int SampleRateHz() const override { return enc_->SampleRateHz(); }
- size_t NumChannels() const override { return enc_->NumChannels(); }
- int RtpTimestampRateHz() const override { return enc_->RtpTimestampRateHz(); }
- size_t Num10MsFramesInNextPacket() const override {
- return enc_->Num10MsFramesInNextPacket();
- }
- size_t Max10MsFramesInAPacket() const override {
- return enc_->Max10MsFramesInAPacket();
- }
- int GetTargetBitrate() const override { return enc_->GetTargetBitrate(); }
- EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
- rtc::ArrayView<const int16_t> audio,
- rtc::Buffer* encoded) override {
- return enc_->Encode(rtp_timestamp, audio, encoded);
- }
- void Reset() override { return enc_->Reset(); }
- bool SetFec(bool enable) override { return enc_->SetFec(enable); }
- bool SetDtx(bool enable) override { return enc_->SetDtx(enable); }
- bool SetApplication(Application application) override {
- return enc_->SetApplication(application);
- }
- void SetMaxPlaybackRate(int frequency_hz) override {
- return enc_->SetMaxPlaybackRate(frequency_hz);
- }
-
- private:
- AudioEncoder* enc_;
-};
-
void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
if (value != last_value_ || first_time_) {
first_time_ = false;
@@ -480,26 +427,6 @@
modifier(&encoder_stack_);
}
-// Get current send codec.
-absl::optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
- rtc::CritScope lock(&acm_crit_sect_);
- if (encoder_stack_) {
- CodecInst ci;
- ci.channels = encoder_stack_->NumChannels();
- ci.plfreq = encoder_stack_->SampleRateHz();
- ci.pacsize = rtc::CheckedDivExact(
- static_cast<int>(encoder_stack_->Max10MsFramesInAPacket() * ci.plfreq),
- 100);
- ci.pltype = -1; // Not valid.
- ci.rate = -1; // Not valid.
- static const char kName[] = "external";
- memcpy(ci.plname, kName, sizeof(kName));
- return ci;
- } else {
- return absl::nullopt;
- }
-}
-
void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
rtc::CritScope lock(&acm_crit_sect_);
if (encoder_stack_) {
@@ -748,54 +675,10 @@
receiver_.SetCodecs(codecs);
}
-bool AudioCodingModuleImpl::RegisterReceiveCodec(
- int rtp_payload_type,
- const SdpAudioFormat& audio_format) {
+absl::optional<std::pair<int, SdpAudioFormat>>
+ AudioCodingModuleImpl::ReceiveCodec() const {
rtc::CritScope lock(&acm_crit_sect_);
- RTC_DCHECK(receiver_initialized_);
-
- if (!acm2::RentACodec::IsPayloadTypeValid(rtp_payload_type)) {
- RTC_LOG_F(LS_ERROR) << "Invalid payload-type " << rtp_payload_type
- << " for decoder.";
- return false;
- }
-
- return receiver_.AddCodec(rtp_payload_type, audio_format);
-}
-
-int AudioCodingModuleImpl::RegisterExternalReceiveCodec(
- int rtp_payload_type,
- AudioDecoder* external_decoder,
- int sample_rate_hz,
- int num_channels,
- const std::string& name) {
- rtc::CritScope lock(&acm_crit_sect_);
- RTC_DCHECK(receiver_initialized_);
- if (num_channels > 2 || num_channels < 0) {
- RTC_LOG_F(LS_ERROR) << "Unsupported number of channels: " << num_channels;
- return -1;
- }
-
- // Check if the payload-type is valid.
- if (!acm2::RentACodec::IsPayloadTypeValid(rtp_payload_type)) {
- RTC_LOG_F(LS_ERROR) << "Invalid payload-type " << rtp_payload_type
- << " for external decoder.";
- return -1;
- }
-
- return receiver_.AddCodec(-1 /* external */, rtp_payload_type, num_channels,
- sample_rate_hz, external_decoder, name);
-}
-
-// Get current received codec.
-int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const {
- rtc::CritScope lock(&acm_crit_sect_);
- return receiver_.LastAudioCodec(current_codec);
-}
-
-absl::optional<SdpAudioFormat> AudioCodingModuleImpl::ReceiveFormat() const {
- rtc::CritScope lock(&acm_crit_sect_);
- return receiver_.LastAudioFormat();
+ return receiver_.LastDecoder();
}
// Incoming packet from network parsed and ready for decode.
@@ -902,10 +785,6 @@
return true;
}
-int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) {
- return receiver_.RemoveCodec(payload_type);
-}
-
int AudioCodingModuleImpl::EnableNack(size_t max_nack_list_size) {
return receiver_.EnableNack(max_nack_list_size);
}
@@ -951,52 +830,4 @@
return new AudioCodingModuleImpl(config);
}
-int AudioCodingModule::NumberOfCodecs() {
- return static_cast<int>(acm2::RentACodec::NumberOfCodecs());
-}
-
-int AudioCodingModule::Codec(int list_id, CodecInst* codec) {
- auto codec_id = acm2::RentACodec::CodecIdFromIndex(list_id);
- if (!codec_id)
- return -1;
- auto ci = acm2::RentACodec::CodecInstById(*codec_id);
- if (!ci)
- return -1;
- *codec = *ci;
- return 0;
-}
-
-int AudioCodingModule::Codec(const char* payload_name,
- CodecInst* codec,
- int sampling_freq_hz,
- size_t channels) {
- absl::optional<CodecInst> ci = acm2::RentACodec::CodecInstByParams(
- payload_name, sampling_freq_hz, channels);
- if (ci) {
- *codec = *ci;
- return 0;
- } else {
- // We couldn't find a matching codec, so set the parameters to unacceptable
- // values and return.
- codec->plname[0] = '\0';
- codec->pltype = -1;
- codec->pacsize = 0;
- codec->rate = 0;
- codec->plfreq = 0;
- return -1;
- }
-}
-
-int AudioCodingModule::Codec(const char* payload_name,
- int sampling_freq_hz,
- size_t channels) {
- absl::optional<acm2::RentACodec::CodecId> ci =
- acm2::RentACodec::CodecIdByParams(payload_name, sampling_freq_hz,
- channels);
- if (!ci)
- return -1;
- absl::optional<int> i = acm2::RentACodec::CodecIndexFromId(*ci);
- return i ? *i : -1;
-}
-
} // namespace webrtc